| /* | 
 |  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 | #ifndef CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ | 
 | #define CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ | 
 |  | 
 | #include <memory> | 
 |  | 
 | #include "call/rtp_demuxer.h" | 
 | #include "call/rtp_stream_receiver_controller_interface.h" | 
 | #include "rtc_base/critical_section.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | class RtpPacketReceived; | 
 |  | 
 | // This class represents the RTP receive parsing and demuxing, for a | 
 | // single RTP session. | 
 | // TODO(nisse): Add RTCP processing, we should aim to terminate RTCP | 
 | // and not leave any RTCP processing to individual receive streams. | 
 | // TODO(nisse): Extract per-packet processing, including parsing and | 
 | // demuxing, into a separate class. | 
 | class RtpStreamReceiverController | 
 |     : public RtpStreamReceiverControllerInterface { | 
 |  public: | 
 |   RtpStreamReceiverController(); | 
 |   ~RtpStreamReceiverController() override; | 
 |  | 
 |   // Implements RtpStreamReceiverControllerInterface. | 
 |   std::unique_ptr<RtpStreamReceiverInterface> CreateReceiver( | 
 |       uint32_t ssrc, | 
 |       RtpPacketSinkInterface* sink) override; | 
 |  | 
 |   // Thread-safe wrappers for the corresponding RtpDemuxer methods. | 
 |   bool AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) override; | 
 |   size_t RemoveSink(const RtpPacketSinkInterface* sink) override; | 
 |  | 
 |   // TODO(nisse): Not yet responsible for parsing. | 
 |   bool OnRtpPacket(const RtpPacketReceived& packet); | 
 |  | 
 |  private: | 
 |   class Receiver : public RtpStreamReceiverInterface { | 
 |    public: | 
 |     Receiver(RtpStreamReceiverController* controller, | 
 |              uint32_t ssrc, | 
 |              RtpPacketSinkInterface* sink); | 
 |  | 
 |     ~Receiver() override; | 
 |  | 
 |    private: | 
 |     RtpStreamReceiverController* const controller_; | 
 |     RtpPacketSinkInterface* const sink_; | 
 |   }; | 
 |  | 
 |   // TODO(nisse): Move to a TaskQueue for synchronization. When used | 
 |   // by Call, we expect construction and all methods but OnRtpPacket | 
 |   // to be called on the same thread, and OnRtpPacket to be called | 
 |   // by a single, but possibly distinct, thread. But applications not | 
 |   // using Call may have use threads differently. | 
 |   rtc::CriticalSection lock_; | 
 |   RtpDemuxer demuxer_ RTC_GUARDED_BY(&lock_); | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_ |